Remove the preprocessor symbol WEBRTC_CODEC_PCM16 (it was always defined)

webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 997 matching lines @@
 void AudioCodingImpl::GetDecodingCallStatistics(
     AudioDecodingCallStats* call_stats) const {
   acm_old_->GetDecodingCallStatistics(call_stats);
 }
 
 bool AudioCodingImpl::MapCodecTypeToParameters(int codec_type,
                                                std::string* codec_name,
                                                int* sample_rate_hz,
                                                int* channels) {
   switch (codec_type) {
-#ifdef WEBRTC_CODEC_PCM16
     case acm2::ACMCodecDB::kPCM16B:
       *codec_name = "L16";
       *sample_rate_hz = 8000;
       *channels = 1;
       break;
     case acm2::ACMCodecDB::kPCM16Bwb:
       *codec_name = "L16";
       *sample_rate_hz = 16000;
       *channels = 1;
       break;
@@ skipping 10 matching lines @@
     case acm2::ACMCodecDB::kPCM16Bwb_2ch:
       *codec_name = "L16";
       *sample_rate_hz = 16000;
       *channels = 2;
       break;
     case acm2::ACMCodecDB::kPCM16Bswb32kHz_2ch:
       *codec_name = "L16";
       *sample_rate_hz = 32000;
       *channels = 2;
       break;
-#endif
 #if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX))
     case acm2::ACMCodecDB::kISAC:
       *codec_name = "ISAC";
       *sample_rate_hz = 16000;
       *channels = 1;
       break;
 #endif
 #ifdef WEBRTC_CODEC_ISAC
     case acm2::ACMCodecDB::kISACSWB:
       *codec_name = "ISAC";
@@ skipping 79 matching lines @@
       *channels = 1;
       break;
 #endif
     default:
       FATAL() << "Codec type " << codec_type << " not supported.";
   }
   return true;
 }
 
 }  // namespace webrtc